This invention relates generally to digital telephone networks, and in particular to techniques for transporting Dual Tone Multi Frequency (DTMF) digits over packet networks.
The Internet and other packet switched networks are increasingly used as a transmission medium for voice telephone calls. Internet telephony software and services now provide low cost, or even free, telephone calls anywhere in the world. With simple equipment at the subscriber end, a virtual connection can be established between two callers through a system of interconnected packet-based networks that may include the Internet, intranets or other digital networks. The Internet is thus emerging as a viable alternative to legacy analog, circuit switched networks, as long as users can tolerate occasional delays and sometimes inferior quality of service.
In order to facilitate communication over the Internet, various industry and international standards bodies have established different functional requirements and rules that govern transmission of data packets. Implementation of these common rules, known as “protocols”, is necessary to allow equipment provided by different manufacturers to inter-operate.
One typical device within a packet network is a so-called gateway. Gateways allow dissimilar computer networks that might use different protocols to connect with one another. A gateway provides, in effect, an interface that translates data between the different communication protocols used. One type of gateway is an Internet Protocol (IP) telephony gateway. A typical IP gateway designed to handle telephone calls can handle multiple simultaneous calls from standard telephone connections originating within the Public Switched Telephone Network (PSTN), and route them over packet networks such as the Internet.
In a typical Voice over Internet Protocol (VoIP) connection, a caller located at an origin point places a telephone call using a standard telephone or computer modem. The call is then routed to a local “originating” Internet telephony gateway which is connected to the Internet. The originating gateway then establishes one or more Internet “sessions” with a remote or “terminating” gateway that services the telephone at the other end of the call. The terminating gateway then completes the circuit by connecting to the destination telephone via a local circuit switched network connection.
In order to communicate voice audio signals in an Internet-based telephone system, the gateways operate on audio signals received from and transmitted to the parties' telephones. These audio signals are typically digital Pulse Code Modulated (PCM) signals that may be formulated according to various standards.
At the origin point, equipment is used to sample digitize and encode an analog voice signal. The encoded bits are then arranged into packets for transmission over the packet networks that provide the virtual connection. At the termination point, other equipment dissembles the packets, decodes the sample bits, and converts them back to an analog voice signal again.
One transport protocol often used for carrying VoIP voice packets between gateways is the Internet Engineering Task Force (IETF) Real Time Transport Protocol (RTP), as defined in Request for Comment (RFC) 1889. RFC 1889 has now been placed into the International Telecommunications Union's (ITU) standard H.225.0.
There has been a challenge, however, in determining how best to carry push button tones, technically referred to as Dual Tone, Multi-Frequency (DTMF) digits. DTMF digits are typically generated as sequences of sine waves, either added or modulated on the voice signal. DTMF digits are now almost universally used as dialed number digits to establish a telephone call connection. Correct transmission of DTMF digits is also important to a caller, however, even after the end-to-end connection is made. For example, the operation of voice mail and other systems such as Interactive Voice Response (IVR) systems are heavily reliant on correct reception of DTMF digits.
In a VoIP connection that uses the RTP protocol, each end of an RTP trunk typically encodes the voice samples with an appropriate coding scheme, such as the so-called G.711, G.723 or G.729 codecs. Equipment that uses linear codecs such as G.711 do not pose a problem since they can faithfully pass the DTMF tones end to end. However, non-linear codecs such as G.723 and G.729 introduce compression to the digital samples. Therefore, these codecs do not pass DTMF tones reliably.
Thus, a gateway presently has two options for handling DTMF digits. First, it can use only a linear codec and make no attempt to handle DTMF tones differently from voice samples. However, when compression codecs are desirable, an originating gateway can detect and recognize an individual DTMF digit and translate it into a data value. The data value can then be encoded into a special type of packet. Upon receipt of this packet at the terminating point, the receiver can then reproduce the corresponding DTMF tone signal.
For example, DTMF tones and other named telephony events (such as modem tones, fax tones, etc.) can be generated packets containing as data values, as described in the proposed standard known as RFC 2833. RFC 2833 is an IETF “standards track” proposal for carrying DTMF digits as RTP packets. According to this standard, the packet includes a data value indicating the particular DTMF digit, as well as a volume and a duration for each DTMF digit.
A time duration of from 60 through 80 milliseconds (ms) has historically been considered to be the minimum sufficient for reliable transport of DTMF tones through a circuit switched network. However, a common expectation is now that originating equipment can transmit a minimum DTMF tone duration of as little as 50 ms, and that receiving equipment should be capable of detecting any DTMF tone of at least 45 ms. Such requirements are, for example, promulgated by the International Telecommunications Union (ITU) in the Q.24 specification.
Accommodation of these shorter DTMF tone durations has not necessarily been a problem in circuit switched networks which can faithfully reproduce short tones via their dedicated end to end connections. This also does not typically pose a problem for the initial DTMF digits that make up a dialed telephone number, since those are not used by gateways.
Such relatively short duration DTMF tones do become a problem when they are needed during a call in progress, such as to control advanced voice applications like voice mail, telephone banking systems, and other Integrated Voice Response (IVR) systems located at the other end of a VoIP connection. These systems are intended to be controllable by a user through a telephone handset after the initial end-to-end connection is made. Once connected, a user is typically presented with menus that are to be navigated by sending DTMF tones from the user handset. Problems may occur, however, when using such systems if an intermediate network, such a VoIP network, introduces the possibility of cutting off such tones.
For example, an originating VoIP gateway might take 5 to 10 ms to detect a DTMF tone. This delay in detection time may introduce an error into the duration measurement. Thus, a DTMF tone which was originally received with a duration of 45 ms may be detected at the originating gateway as only being between 35 and 40 ms long. This in turn might cause some equipment to miss a DTMF digit.
An additional difficulty is presented by the fact that it is not always possible to playback tones at the destination with the same duration originally created at the origin point. Thus, even if the tone duration can be correctly reproduced, other impairments in the network introduce an additional complication.